[..] |
ASTERISK-15000: Phone init problem solved with backport. |
ASTERISK-15001: chan_sip breaks RFC by incrementing session version between non-reliable 1xx and 200 |
ASTERISK-15002: Crash on many connected / canceled calls |
ASTERISK-15003: [patch] Add couple of useful extensions as examples |
ASTERISK-15004: [patch] Security Problem |
ASTERISK-15005: Crash in chan_dahdi with ss7 support |
ASTERISK-15006: SIP subscriptions are lost after a reload |
ASTERISK-15007: [patch] iax2 show cache, locks channels. |
ASTERISK-15008: With delayreject=yes calls sometimes fail |
ASTERISK-15009: Exceptionally long voice queue length queuing to Local/XXXXXXX |
ASTERISK-15010: [patch] asterisk crashes when there are no RTP port left |
ASTERISK-15011: Can't dial to exchange UM 2007 |
ASTERISK-15012: [regression] 1.4 SVN branch does not write Userfield in CDR |
ASTERISK-15013: [patch] default say.conf for new number method doesnt handle all numbers |
ASTERISK-15014: Sometimes macro in h extension returns to s extension |
ASTERISK-15015: app_voicemail appending IMAPFOLDER to 'vm-' to create filename for prompt to play. |
ASTERISK-15016: [patch] incorrect playback when using say_date_with_format_es on one o'clock (spanish) |
ASTERISK-15017: [patch] Hangup extension executed twice in 1.6.2 RC2 |
ASTERISK-15018: Kapanga softphone exposes bug in SIP channel driver |
ASTERISK-15019: Application Extenspy |
ASTERISK-15020: Cant delete temporary greetings |
ASTERISK-15021: [patch] Auto-fallthrough when attempting to enter DTMF using Background() in a Macro() |
ASTERISK-15022: Lockup in chan_sip |
ASTERISK-15023: [patch] Fix/improve transaction/dialog-matching in pedantic mode |
ASTERISK-15024: G726 is choppy on IAX - 1.6 |
ASTERISK-15025: ExternalIVR without argument causes segmentation fault |
ASTERISK-15026: SIP peers are not being built from users.conf configuration |
ASTERISK-15027: [patch] ExternalIVR TCP client functionality does not work |
ASTERISK-15028: RTP Media Port Change Ignored |
ASTERISK-15029: [patch] ExternalIVR trapping non-existent files does not work |
ASTERISK-15030: Trying to send a fax with zoiper (t38) causes "buffer overflow message too long" |
ASTERISK-15031: new libpri features not detected with a custom libpri location |
ASTERISK-15032: hylafax(+iaxmodem)+ReceiveFAX leads to "Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED" |
ASTERISK-15033: overlap dial from BRI phone: unlimited number of digits |
ASTERISK-15034: flags not initalized in app_softhangup, causes undefined behavoir |
ASTERISK-15035: Asterisk crashes randomly with a segmentation fault in __res_vinit () |
ASTERISK-15036: [patch] Service indicator not managed |
ASTERISK-15037: ChanSpy crashes Asterisk |
ASTERISK-15038: spaces in caller id name cause unexpected behavior |
ASTERISK-15039: [patch] ast_gethostbyname doesn't set h_length if argument is an IP Address |
ASTERISK-15040: Meetme - Quitting time issue |
ASTERISK-15041: [patch] ExternalIVR Does not use copy IP Address correctly |
ASTERISK-15042: [patch] realtime function does not return pair when database value is null |
ASTERISK-15043: CVE-2008-7220: static-http/prototype.js is vulnerable to "cross-site ajax requests" |
ASTERISK-15044: hint not updated correctly on outgoing SIP calls |
ASTERISK-15045: IAX trunk clicks as other calls in the same trunk hang up |
ASTERISK-15046: Literal values wrapped in documentation |
ASTERISK-15047: Need a CLI command to force a reconnect of ODBC connections |
ASTERISK-15048: [patch] ExternalIVR does not return event for file when file playing is interrupted |
ASTERISK-15049: Callerid is not recorded in database |
ASTERISK-15050: MOH silence |
ASTERISK-15051: Unable to create/find SIP channel for this INVITE |
ASTERISK-15052: [patch] MixMonitor thread doesn't exit until channel is dropped |
ASTERISK-15053: [patch] Extend slin16 support to SIP calls |
ASTERISK-15054: hangup on transfer |
ASTERISK-15055: [patch] Use pkg-config to find gmime libraries. |
ASTERISK-15056: Asterisk core-dumps if used as loadgenerator using callfiles |
ASTERISK-15057: [patch] Wrong cause value for 'answered elsewhere' |
ASTERISK-15058: [regression] High CPU usage, choppy sound |
ASTERISK-15059: Calltoken and Realtime |
ASTERISK-15060: Crash on meetme leave |
ASTERISK-15061: [patch] incorrect 'core show channel channel-name' output |
ASTERISK-15062: asterisk continiously crashes when iax-call received |
ASTERISK-15063: [patch] Limit on simultaneous incoming calls for queue members |
ASTERISK-15064: [patch] Setting dialplan hint and using a global variable gives incorrect warning. |
ASTERISK-15065: Chaspy always active Option ´o´ |
ASTERISK-15066: full system crash every other day |
ASTERISK-15067: [patch] temporary greetings can't be erased in 1.4 |
ASTERISK-15068: AgentComplete event sent as soon as call is answered in call to queue through local channel |
ASTERISK-15069: [patch] "make config" creates really wrong runlevels in Debian (includes patch) |
ASTERISK-15070: [patch] chan_mgcp new feature: digitmaps definitions |
ASTERISK-15071: [patch] Asterisk does not fully support SIP connections to Internet Telephony Service Providers |
ASTERISK-15072: Revision 202007 Introduces Deadlock |
ASTERISK-15073: OpenSolaris Build Problem with editline |
ASTERISK-15074: Crash in SQLAllocHandle |
ASTERISK-15075: res_pktccops.c using MSG_NOSIGNAL |
ASTERISK-15076: Asterisk Crash after SIP Transfer |
ASTERISK-15077: [regression] Whisper mode in ChanSpy() has delays and gaps in audio (sometimes not working at all) |
ASTERISK-15078: Dial Option D() does not execute in parallel witb option A() |
ASTERISK-15079: [patch] Thousands of Invites never discarded in sip channels |
ASTERISK-15080: [patch] out of dialog SIP_NOTIFY with event='keep-alive' |
ASTERISK-15081: Inband onhook ringing not applying indications.conf settings |
ASTERISK-15082: Crash of Outgoing Call |
ASTERISK-15083: [patch] T.38 reinvite fails after receiving "415 Unsupported media type" when it could continue in audio mode |
ASTERISK-15084: [patch] RFC 4474 Implementation of SIP identity |
ASTERISK-15085: chan_mobile pairs, dials, and receives calls, but no audio |
ASTERISK-15086: [patch] if, for, while, switch statements all missing space, - Coding guidelines |
ASTERISK-15087: chan_mobile.c needs updating for 32->64 bit changes |
ASTERISK-15088: [patch] Segfault with limit data L(x:y) and verbosity >= 3 |
ASTERISK-15089: IAX2 Codec negotiation fails since 227580 |
ASTERISK-15090: Can't compile H323 channel driver |
ASTERISK-15091: Core dump in audio_audiohook_write_list |
ASTERISK-15092: Core dump in vsnprintf / ast_rtp_get_quality / sip_hangup |
ASTERISK-15093: WARNING channel.c __ast_channel_alloc_ap: Channel allocation failed: Can't create alert pipe! |
ASTERISK-15094: Outbound proxy with realtime integration not working |
ASTERISK-15095: sound files missing for say_enumeration: digits/h-hundred |
ASTERISK-15096: T.38 Fax Termination Failing |
ASTERISK-15097: [patch] Asterisk 1.6.0.13 Asterisk crashes when running dialplan app macro on a macro that does not exist. |
ASTERISK-15098: [patch] Backport patch 9048 to v1.4: Provide colors for daemonized asterisk |
ASTERISK-15099: Call failed to go through, reason (8) Congestion (circuits busy) Description: When launching more t |
ASTERISK-15100: Asterisk Freezes when more than 5 simultaneous calls using chan_mobile |
ASTERISK-15101: [patch] Segfault in chan_iax2.so when receiving call without CallToken support |
ASTERISK-15102: [patch] asterisk keeps starting new processes for streaming audio MOH |
ASTERISK-15103: Call Hold not working on some phones |
ASTERISK-15104: [patch] When using ooh323h it is impossible to call when number have more than 3 number... |
ASTERISK-15105: Setting a call-forward on an analog phone results in the analog phone still being rung when dialed |
ASTERISK-15106: [patch] chan_ooh323 don't works with Avaya Definity |
ASTERISK-15107: [patch] Event collision in ExternalIVR resolved by documenting issue |
ASTERISK-15108: [regression] Early audio message doesn't play over SIP |
ASTERISK-15109: [patch] Script to automatically email backtrace to admin |
ASTERISK-15110: A func_math WARNING for a operation that is not executed in ExecIf |
ASTERISK-15111: [patch] Invalid behaviour of Return within Gosub and AGI |
ASTERISK-15112: [patch] #define MAX_LANGUAGE increment from 20 to 30 in include/asterisk/channel.h |
ASTERISK-15113: Asterisk does not connect the call to internal extension |
ASTERISK-15114: Crash revision 229091 in audiohook_inheritance_destroy |
ASTERISK-15115: [patch] Fix ExternalIVR Documentation in 1.4 |
ASTERISK-15116: T.38 passthrough issue in 1.6.1.6 |
ASTERISK-15117: Undefined references in function `agent_set_base_channel': |
ASTERISK-15118: [patch] CDR always set disposition as NO ANSWER with .call files |
ASTERISK-15119: [patch] "requirecalltoken" config directive not respected globally |
ASTERISK-15120: [patch] Remove features from ExternalIVR documentation |
ASTERISK-15121: 1.4.26.3 security issue - Chinese IPs somehow are making calls without authentication |
ASTERISK-15122: Call files produces "NO ANSWER" record |
ASTERISK-15123: eswitch does not substiotute variables when using Local Channel |
ASTERISK-15124: ERROR[24164]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
ASTERISK-15125: [patch] g726 to g726aal2 translation and cost-calculation are wrong but easily fixed. |
ASTERISK-15126: Failed assertion in chan_iax2 update_registry/ast_sched_del |
ASTERISK-15127: [patch] Call parking via AMI causes announcment and ringback to caller channel |
ASTERISK-15128: Placing a URI call fails when URI string contains a non-standard port |
ASTERISK-15129: Can't initiate outbound SIP calls when siren14 is the ONLY enabled codec |
ASTERISK-15130: [patch] Building Queues with Asterisk - A How-to Guide |
ASTERISK-15131: sip calls drop because of BYE's |
ASTERISK-15132: AMD() incorrectly sets AMDCAUSE channel variable |
ASTERISK-15133: [patch] QUEUE_MEMBER(...,free) counts wrapping-up agents as available |
ASTERISK-15134: [patch] issues in processing "Action: Events" eventmask |
ASTERISK-15135: php agi causes segfaults in php -> /var/lib/asterisk/agi-bin/fixlocalprefix |
ASTERISK-15136: [patch] Iconv is a glibc-ism, and as such should be called out explicitly |
ASTERISK-15137: [patch] [regression] The status of External SIP peer used as Queue member is not updating correctly |
ASTERISK-15138: ast_rtp_destroy causes segmentation violation |
ASTERISK-15139: [patch] Muted user remains talking forever |
ASTERISK-15140: detect fsk caller-id with DT-AS start signal |
ASTERISK-15141: UDPTL asked to send 77 bytes of IFP when far end only prepared to accept 45 bytes; data loss may occur. |
ASTERISK-15142: Symbol referencing errors (MIN/MAX in channel.o/udptl.o) |
ASTERISK-15143: ODBC Crash in 1.6.0.18-rc2 |
ASTERISK-15144: hello |
ASTERISK-15145: SIP REFER initiated from the Asterisk Transfer application fails on a SOPHO iS3000 SIP Server |
ASTERISK-15146: DAHDIScan() only returns dead air |
ASTERISK-15147: MixMonitor stops audio on native bridging |
ASTERISK-15148: [patch] Memory leak in res_config_ldap when using realtime extensions |
ASTERISK-15149: Missing CDR |
ASTERISK-15150: core dump somewhere in format_wav or res_musiconhold (1.4.27-RC2) |
ASTERISK-15151: [patch] Allow execincludes within asterisk.conf |
ASTERISK-15152: [patch] Conditional jump or move depends on uninitialised STACK value |
ASTERISK-15153: ChanSpy Whisper not working properly when peer has VAD and CNG on. |
ASTERISK-15154: [patch] VM_DATE does not follow emaildateformat format for pager email |
ASTERISK-15155: [patch] UserEvent manager action is not ACKed |
ASTERISK-15156: Etisalat UAE Disconnect Tone |
ASTERISK-15157: [patch] Trunk won't build as cross-compilation |
ASTERISK-15158: [patch] unanswered has no effect |
ASTERISK-15159: [patch] Last line of SDP is not being parsed |
ASTERISK-15160: incomming call on agent when an agent is in outgoing call. |
ASTERISK-15161: [patch] Asterisk doesn't free udp ports |
ASTERISK-15162: [patch] message limit (maxmsg) can be exceeded in 1.6.x creating orphan voicemail |
ASTERISK-15163: [patch] Language code collisions for certan languages |
ASTERISK-15164: crash and core dump |
ASTERISK-15165: Asterisk 1.6.0.13 Asterisk crashes intermittently cause unknown. |
ASTERISK-15166: [patch] response to "Action: Events" is not finished by empty line |
ASTERISK-15167: No way to set pin for new ConfBridge conferences. |
ASTERISK-15168: Outoing calls disconnected immediately after remote end picks up. |
ASTERISK-15169: [patch] Incoming multiline SMS causes chan_mobile to stop working |
ASTERISK-15170: [patch] asterisk reload causes mpg123 streams to be recreated |
ASTERISK-15171: RTPAUDIOQOS and RTPAUDIOQOSBRIDGED false statistics |
ASTERISK-15172: Unable to negotiate codecs using IAX2 (and possibly others) |
ASTERISK-15173: Double CDR if unanswered call |
ASTERISK-15174: somtimes when agent is at conversation with queue caller, call disconnected and new person begin his conversation |
ASTERISK-15175: [patch] Support for disabling automon selectively per peer |
ASTERISK-15176: Review of internal_timing code |
ASTERISK-15177: Implement a pin auth for ConfBridge, like conferences in meetme.conf have |
ASTERISK-15178: G723 codec has digitzed voice |
ASTERISK-15179: [patch] core show function CDR reports wrong disposition values |
ASTERISK-15180: ast_ouraddrfor doesn't do htons() on the port |
ASTERISK-15181: app_voicemail.c strip_control() strips more than just control chars |
ASTERISK-15182: [patch] [regression] Asterisk sip.conf realtime register, contact problem |
ASTERISK-15183: [patch][regression] DTMF Not Recognized with Exchange UM |
ASTERISK-15184: [patch] G.719 Pass-through Support for Asterisk |
ASTERISK-15185: Outgoing Caller ID Name For QSIG |
ASTERISK-15186: [patch] handle_incoming() incorrectly sets p->method to SIP_ACK |
ASTERISK-15187: [patch] menuselect.makeopts: does not properly unselect an option with a leading - (minus) |
ASTERISK-15188: [patch] Timeout in SPEECH RECOGNIZE not working. |
ASTERISK-15189: [patch] After upgrading to asterisk 1.4.27 Optipoint SIP phone can no longer register |
ASTERISK-15190: [patch] pedantic sip checking needed to generate valid messages (but broken) |
ASTERISK-15191: 64bit Host OS, 32bit OpenVZ/Virtuozzo VPS, Dahdi does not work |
ASTERISK-15192: Asterisk Reference Information has incomplete coverage of sip.conf file |
ASTERISK-15193: [patch] Asterisk crashes with Asyc AGI when hangup |
ASTERISK-15194: [patch] Some warnings when parsing extensions.conf fail to include line numbers |
ASTERISK-15195: g729 doesnt work when asterisk installed with ([*] LOW_MEMORY) |
ASTERISK-15196: [patch] ExternalIVR confuses AGI by double closing FDs |
ASTERISK-15197: Interlock in chan_dahdi |
ASTERISK-15198: IAX calls drop after ~30 seconds between 1.4.27rc5 and 1.2.36 |
ASTERISK-15199: [regression] Old/new message Seen/Unseen is not RFC compliant |
ASTERISK-15200: segfault in "core show functions" |
ASTERISK-15201: 488 not acceptable when receiving T.38 at 14400 speed |
ASTERISK-15202: Recursion crash in pbx_ael.c |
ASTERISK-15203: dummy_start (data=0xb7c9be00) at utils.c and in clone () from /lib/libc.so.6 |
ASTERISK-15204: Responds sendrecv to recvonly SDP, but RFC 3264 says sendonly and inactive are only possible replies |
ASTERISK-15205: When placing an external SIP trunk caller on hold, no music on hold audio is heard |
ASTERISK-15206: Useless MySQL queries when doing sip qualify |
ASTERISK-15207: [regression] After upgrading i see a lot more Notices peers status |
ASTERISK-15208: [patch] stop the flame - remove 'silly' from channel.c |
ASTERISK-15209: [patch] New SDP handling code totally broke T.38 reinvites |
ASTERISK-15210: SRV registration |
ASTERISK-15211: [patch] When using 'joinempty=strict', "in use" devices not seen as "unavailable". |
ASTERISK-15212: [patch] Incorrect reloading of realtime peer causes mailbox list to expand indefinitely |
ASTERISK-15213: [patch] [regression] asterisk deadlocks and calls will stop queueing. |
ASTERISK-15214: segfault if too many rooms in meetme.conf |
ASTERISK-15215: [patch] confusing description in configs/sip.conf.sample |
ASTERISK-15216: Autodestruct |
ASTERISK-15217: Siemens S685IP g722 gets not translated |
ASTERISK-15218: Asterisk responds 488 - Not acceptable here on T38 reinvite |
ASTERISK-15219: Asterisk drops call leg to Cisco while other leg remains up. |
ASTERISK-15220: chan_sip transforms %23 to # (UTF-8 issue) in Contact field |
ASTERISK-15221: Asterisk is crashing on any H323 call. |
ASTERISK-15222: [patch] A blind transfer results in a call with empty accountcode in a specific circumstance |
ASTERISK-15223: When trying to enable jitter buffer on local channel atserisk crash |
ASTERISK-15224: Interlock between directed pickup and device state threads |
ASTERISK-15225: Segmentation fault in chan_sip in function initreqprep |
ASTERISK-15226: [patch] Asterisk crashes randomly on mISDN RELEASE_COMPLETE |
ASTERISK-15227: reinvites fail when sdp-session does not increment |
ASTERISK-15228: [patch] Segmentation Fault on Originate command. |
ASTERISK-15229: [patch] chan_mobile doesn't hangup |
ASTERISK-15230: RFC 2833 DTMF Events Generated by Polycom IP Phone Running v3.2.1 F/W Not Recognized by Asterisk 1.4 |
ASTERISK-15231: [patch] Fix bootstrap.sh on OpenSolaris |
ASTERISK-15232: [patch] configure fails to detect spandsp/expose.h when not in system include path |
ASTERISK-15233: [patch] [branch] New CLI command: manager show settings |
ASTERISK-15234: pbx_extension_helper cannot find labels in contexts |
ASTERISK-15235: Call terminates 5 seconds after establishing |
ASTERISK-15236: chan_dahdi uses 1-2 for second port on one span |
ASTERISK-15237: Custom CDR values not logged when dialing with local channels |
ASTERISK-15238: [patch] get_sdp_line condition is not right? |
ASTERISK-15239: System completely hangs after executing an ODBC function |
ASTERISK-15240: [patch] Deleting Multiple IMAP voicemails does not work reliably |
ASTERISK-15241: Asterisk 1.6.1 won't "answer" the phone when using a callcentric sip trunk |
ASTERISK-15242: transmit_refer leaks sip_refer structures |
ASTERISK-15243: [patch] keepstats option removed when it shouldn't |
ASTERISK-15244: Segfault in ast_frdup |
ASTERISK-15245: [patch] Hints do not have the correct state on initialization |
ASTERISK-15246: Interlock between SIP and device state threads |
ASTERISK-15247: [patch] Send Manager Event on SNOM X-ClientCode SIP INFO message |
ASTERISK-15248: [patch] new option: lockconfdir for protecting conf files in /etc/asterisk during reloads |
ASTERISK-15249: [patch] Only the last setvar is effective for a given channel |
ASTERISK-15250: LOCK behaves like trylock (not waiting for 3 seconds) |
ASTERISK-15251: [patch] Asterisk crashes after receiving fax with 'double free' |
ASTERISK-15252: The current SVN does not compile |
ASTERISK-15253: [patch] Asterisk crash when SpeechCreate() is used in dialplan without exact name of the module, using the default. |
ASTERISK-15254: [patch] Since changeset 231437: Queue ERROR[7429]: astobj2.c:114 INTERNAL_OBJ: bad magic number 0x0 for 0xb7174c50 |
ASTERISK-15255: SendFax sessions not correctly reported |
ASTERISK-15256: Asterisk detects DTMF inband even when dtmfmode=rfc2833 |
ASTERISK-15257: Asterisk crash in rtp.c a few times a day |
ASTERISK-15258: Asterisk ignoring sendonly SDP generated from Cisco UCM after generating inactive SDP when a Cisco phone initiates hold |
ASTERISK-15259: Crash due to fault about twice daily |
ASTERISK-15260: Crash when performing dial |
ASTERISK-15261: [patch] Incorrect path passed to MONITOR_EXEC application after 'Monitor()' call finishes. |
ASTERISK-15262: thereis no sample for asterisk.conf |
ASTERISK-15263: res_monitor.c chan->monitor->filename_base has duplicated path |
ASTERISK-15264: [patch] mpg123 <defunct> |
ASTERISK-15265: [patch] Patch: New admin features: Roll call, eject all, mute all, record in-conf |
ASTERISK-15266: [patch] Implicit declaration of 'ast_complete_source_filename' and 'ast_rtp_destroy' with LOW_MEMORY enabled in trunk |
ASTERISK-15267: About issue-0012950 [patch] PacketCable NCS 1.0 Support for Docsis / Eurodocsis Networks.. |
ASTERISK-15268: [patch] Add support for ring indication when calling member |
ASTERISK-15269: [patch] app_queue segfaults if realtime field uniqueid is NULL |
ASTERISK-15270: [patch] Missing session level connection data (c=) breaks process_sdp() |
ASTERISK-15271: [patch] New music on hold patches cause asterisk + full system hard lock |
ASTERISK-15272: [patch] busydetect incorrectly hangs up incoming call due to incoming DTMF seen as busy pattern. |
ASTERISK-15273: Asterisk 1.6.1.9 lockup when caller hangs up in StartMusicOnHold() |
ASTERISK-15274: [patch] App MeeMe Set the channels' account code to the conference room number |
ASTERISK-15275: vsendonly is write only |
ASTERISK-15276: Build fails on OpenBSD4.2 in utils.o |
ASTERISK-15277: [patch] Segfault in res_config_ldap |
ASTERISK-15278: Error in ulaw |
ASTERISK-15279: [patch] SIP Realtime SQL Table |
ASTERISK-15280: [patch] rtpkeepalive parsed twice |
ASTERISK-15281: [patch] cdr_mysql driver does not have an option to log in GMT time |
ASTERISK-15282: dahdi show channels does not show an outgoing call |
ASTERISK-15283: [patch] New option setvarout which sets channel variable for outbound channels to a peer |
ASTERISK-15284: [patch] Missing \n in logging |
ASTERISK-15285: segfault error 4 in libpthread on Ubuntu |
ASTERISK-15286: [patch] potential buffer overflow in say_date_with_format() |
ASTERISK-15287: Asterisk runs out of handles because of stuck SIP dialogs |
ASTERISK-15288: [patch] First DTMF digit is missed if pressed during "using your touch tone keypad..." announcement |
ASTERISK-15289: [patch] IP and port is not transferred for t.38 |
ASTERISK-15290: Ignoring unknown format wav & wav49... |
ASTERISK-15291: T.38 Fax Termination Failing Resolution -- Request for More Information |
ASTERISK-15292: [patch] Message forwarding with prepention does not backup original message and length as intended |
ASTERISK-15293: [patch] Portability tweaks to contrib/scripts/safe_asterisk |
ASTERISK-15294: ChanSpy and ExtenSpy applications don't accept a colon-delimited list of groups |
ASTERISK-15295: one way audio after call waiting |
ASTERISK-15296: Asterisk 1.6.2: CDR is not produced with .call files |
ASTERISK-15297: Asterisk ignore SIP signalling path |
ASTERISK-15298: SIP qualify fails unless NAT is enabled |
ASTERISK-15299: [patch] remainder ast_expr2 func misspelt as reminder |
ASTERISK-15300: [patch] park() function takes 100% of CPU |
ASTERISK-15301: [patch] Callee on outside line can take parking and forwarding rights |
ASTERISK-15302: Asterisk crashes on dtmf detection on channel with 2 bluetooth cellphone |
ASTERISK-15303: [patch] Send manager event on Call Pickup |
ASTERISK-15304: racecondition leading to deadlock in chan_local |
ASTERISK-15305: app_voicemail will module will not load |
ASTERISK-15306: [patch] Background() when called from AGI script no longer gives digit code when interrupted |
ASTERISK-15307: Crash when making outbound call |
ASTERISK-15308: asterisk is not able to register with SIP server |
ASTERISK-15309: [patch] Unable to escape back to dialplan or operator, using 'o' and 'a' extensions in dialcontext |
ASTERISK-15310: [regression] DTMF Tones not working |
ASTERISK-15311: [patch] silencethreshold=0 when dsp.conf not existing |
ASTERISK-15312: Can't handle frames in 2 format - revisited |
ASTERISK-15313: chan sip removes peers like if srvlookup were active, but it is not |
ASTERISK-15314: Asterisk wrongfully sends 403 instead of 401 |
ASTERISK-15315: segfault error 4 |
ASTERISK-15316: Voicemail messages flagged as urgent do not get emailed |
ASTERISK-15317: Trunk does not compile (again) on Darwin (MacOS 10.5) |
ASTERISK-15318: .call file does not create cdr |
ASTERISK-15319: core show hints do not follow the general sorting ordre |
ASTERISK-15320: [patch] Lots of crashes after upgrading to latest 1.6.0.20-rc1 |
ASTERISK-15321: the value of odbcstorage is not taken into account |
ASTERISK-15322: [patch] ASTARGS in sysconfig not inherited as startup options |
ASTERISK-15323: [patch] Manager hooks don't execute if there aren't any manager sessions |
ASTERISK-15324: IAX2 Can't compress subclass 4294967295 |
ASTERISK-15325: [Patch] always m=text 0 in sdp answer |
ASTERISK-15326: Variables not Passed |
ASTERISK-15327: [patch] Change in sip show channels display format allowing more digits for CID |
ASTERISK-15328: [patch] DTMF CallerID detection without polarity reversal |
ASTERISK-15329: [patch] for reading and writing to text file |
ASTERISK-15330: [regression] Record application hangs up after exactly 30 seconds, with or without silence or duration specified |
ASTERISK-15331: make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o |
ASTERISK-15332: [patch] Set time for use in evaluating holiday dialplans |
ASTERISK-15333: Asterisk sends port 0 in the Register message to Proxy if listen port is dynamic |
ASTERISK-15334: [patch] ast_filecopy: keep modes for the created file |
ASTERISK-15335: [patch] cidname and cidnum in output of "sip show peers" |
ASTERISK-15336: [patch] missing newline after reply to event manager-action |
ASTERISK-15337: [patch] [regression] Custom devsate set to INUSE but shows as UNAVAILABLE when accessing through the dialplan |
ASTERISK-15338: "sip show peers" returns notice |
ASTERISK-15339: [patch][regression] VMAuthenticate not playing greeting |
ASTERISK-15340: [patch] Serious problem with pattern matching (regression in #15421) |
ASTERISK-15341: ReceiveFAX always end with "Transmission error" but Fax transmitted successfully |
ASTERISK-15342: Aastra phones won't register SIP |
ASTERISK-15343: after udp error sip phones get kicked |
ASTERISK-15344: [patch] Send manager event on Call Forward |
ASTERISK-15345: [regression] Build fails when defs are required by the linker |
ASTERISK-15346: [patch] When setting a soundfile for announce with a length longer then 80 chars a storage overlay happens |
ASTERISK-15347: [patch] Missing plus signs in MAKE/SUBMAKE calls prevent parallel make from operating correctly |
ASTERISK-15348: CLI reports wrong data |
ASTERISK-15349: 1.6.1.12-rc1 crash around 100 SIP call setup with media |
ASTERISK-15350: [patch] utils/extconf.c growing apart from main/pbx.c |
ASTERISK-15351: one peer for SIP provider with SRV |
ASTERISK-15352: [patch] [regression] T.38 no longer functions |
ASTERISK-15353: [patch] Added musiconhold class in manager event |
ASTERISK-15354: No hold event is generated for a second call on a SIP channel |
ASTERISK-15355: [patch] port in users.conf is not honored in the register statement |
ASTERISK-15356: [patch] Some small solaris fixes (threadstorage / moh) |
ASTERISK-15357: Segfault in res_agi with no second paramter to EXEC |
ASTERISK-15358: [patch] Asterisk man page outdated |
ASTERISK-15359: [patch] Segmentation fault using manager http MXML |
ASTERISK-15360: Exceptionally long voice queue length with chan_iax2 + res_timing_pthread causes high CPU usage |
ASTERISK-15361: [regression] app_sms not working in 1.6.1.12 (same as 0012779) |
ASTERISK-15362: [patch] meetme can support only 6341 rooms |
ASTERISK-15363: [patch] Local values not set within gosub |
ASTERISK-15364: [patch] Monitor resumes recording after SIP transfer despite StopMonitor() having been called |
ASTERISK-15365: Asterisk Crashes when a fax comes in over nvfaxdetect |
ASTERISK-15366: rtp.c:2482 ast_rtcp_write_sr: rtcp halted Operation not permitted |
ASTERISK-15367: [patch] Transferee can hear silence on attended transfer when tranferer hangs up (MOH stops to play) |
ASTERISK-15368: Asterisk causes crosstalk between inbound and random channels |
ASTERISK-15369: depreciated minmessage still referred to in warning |
ASTERISK-15370: gsm to ulaw transcoding sounds terrible |
ASTERISK-15371: Segfault while setting up T.38 fax reception |
ASTERISK-15372: segmentation fault |
ASTERISK-15373: Queue with wrapuptime "call" agent that shouldn't have any call |
ASTERISK-15374: No CDR record for non-bridged outgoing calls |
ASTERISK-15375: [patch] Optimize queries to cache matches |
ASTERISK-15376: [regression] MixMonitor stops recording after transfer using AUDIOHOOK_INHERIT |
ASTERISK-15377: When hanging up a channel running chanspy, chanspy does not exit |
ASTERISK-15378: Chanspy cannot spy on a non-bridged channel |
ASTERISK-15379: [patch] Cannot spy on channel when a local channel is involved |
ASTERISK-15380: [patch] Support for SonyEricsson T6x0 and friends is broken |
ASTERISK-15381: Verbose for Call Parking is incorrect |
ASTERISK-15382: [patch] Bridge application fails when both channels have a similar name |
ASTERISK-15383: [patch] sin_family not set to AF_INET when running trunk on Solaris nevada |
ASTERISK-15384: Wrong Caller ID on Sip Channel when have a lot of hanging up call |
ASTERISK-15385: [regression] chan_local audio crash |
ASTERISK-15386: call answer macro not being called in cmd Dial |
ASTERISK-15387: [patch] Realtime is broken, blank strings aren't valid any more |
ASTERISK-15388: mixmonitor CLI command is broken |
ASTERISK-15389: ${BLINDTRANSFER} not set upon transfer |
ASTERISK-15390: Updating field in realtime queue table does not take effect |
ASTERISK-15391: BlackBerry 7290 will not connect |
ASTERISK-15392: memory leak in confic.c |
ASTERISK-15393: [patch] call pickup can pickup caller instead of callee |
ASTERISK-15394: [regression] Asterisk says "No compatible codecs, not accepting this offer!" on T.38 offer |
ASTERISK-15395: Dialout from Meetme conference |
ASTERISK-15396: chan_unistim randomly crashes |
ASTERISK-15397: Local channel not terminated (regression) |
ASTERISK-15398: (local_queue_frame): Error obtaining mutex: Invalid argument (causes crash) |
ASTERISK-15399: [patch][regression] Macro executes "h" extension instead of exiting |
ASTERISK-15400: Conference Calls |
ASTERISK-15401: [patch] TestServer application does not wait long enough after sending its last DTMF digit |
ASTERISK-15402: "minute" sound file missing. used by app_queue |
ASTERISK-15403: Chan_dahdi calls keep on increase until all the dahdi channel full |
ASTERISK-15404: [patch] Huge memory leak |
ASTERISK-15405: [patch] Manager interface 'Masquerade' event doesn't include Unique id fields |
ASTERISK-15406: [patch] Asterisk crashes in ast_rtcp_write at rtp.c:3536 |
ASTERISK-15407: [patch] Asterisk produces malformed email files for voicemail |
ASTERISK-15408: app_mp3 and chan_local fail |
ASTERISK-15409: [patch] Duration and Billsec Decimal Place |
ASTERISK-15410: [regression] Voicemail message not recording when voicemail.conf format=wav|gsm|wav49 |
ASTERISK-15411: Ignores bindaddr on reload |
ASTERISK-15412: Error parsing format= parameter in voicemail.conf |
ASTERISK-15413: [patch] "config reload" doesn't work correctly |
ASTERISK-15414: crash: in "scheduled_destroy" at chan_iax2.c:1511 |
ASTERISK-15415: Contact header port ignores transport when using externip |
ASTERISK-15416: Always get network congestion on second group using .call file |
ASTERISK-15417: [patch] [regression] Voicemail information is repeated |
ASTERISK-15418: res_fax-1.6.1.5_1.1.6 doesn't trigger T.38 reinvites on fax send (receive works) |
ASTERISK-15419: [patch] ParkAndAnnounce() Does Not Seem To Respect Multiple Parking Lots |
ASTERISK-15420: ISDN to SIP doesn't generate SIP 180 Ringing with Call Progress ISDN message |
ASTERISK-15421: SayUnixTime plays nothing if say.conf mode=new and a format is specified |
ASTERISK-15422: [patch] LD (llinker) options not used by main/ and channels/ builds |
ASTERISK-15423: app_jack fails to connect to jackd |
ASTERISK-15424: ooh323 does not support h245alphanumeric dtmf mode |
ASTERISK-15425: [patch] Add Calling and Called Subaddress to CDR record |
ASTERISK-15426: app_sms hangs on a call sending an sms |
ASTERISK-15427: [patch] astcanary does not exit when asterisk 1.6.2 dies => reopen of #14538 |
ASTERISK-15428: [regression] I got warnings on remote server when transmitting iax variables |
ASTERISK-15429: Asterisk does not send "183 Session Progress" when dialing through a dahdi analog line |
ASTERISK-15430: chan_iax2.c dead lock at chan_iax2.c:2563 |
ASTERISK-15431: [patch] ast_event_cmp always return 1. |
ASTERISK-15432: [patch] Deadlock on &(&channels)->lock |
ASTERISK-15433: Record() doesn't produce recorded file on hangup with 'x' option |
ASTERISK-15434: [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller |
ASTERISK-15435: [patch] [OpenSolaris] wav format produces garbage files |
ASTERISK-15436: CDR and ForkCDR write wrong data on hangup in AGI execution |
ASTERISK-15437: Voicemail subscribtion error |
ASTERISK-15438: [patch] option p added to PickupChan to enable pickup a ringing phone by specifying the peer name |
ASTERISK-15439: [patch] Group Variables |
ASTERISK-15440: Added ability to send DTMF from ExternalIVR |
ASTERISK-15441: [patch] Announce to user when they have been muted/unmuted from the AMI |
ASTERISK-15442: [patch] Additional information for MeetmeJoin |
ASTERISK-15443: [regression] pbx_config does not load when it should |
ASTERISK-15444: [patch] Complile issue with h323plus 1.21.0 and pwlib 2.4.5 |
ASTERISK-15445: asteriskgui not be accessable and softohone |
ASTERISK-15446: [patch] overlap dial should terminate when "ast_matchmore_extension" is false |
ASTERISK-15447: [patch] ExtensionState should resolve dynamic hints |
ASTERISK-15448: [regression] soxmerge arguments broken if Monitoring to absolute path |
ASTERISK-15449: RFC2833 DTMF is not passed correctly when going SIP->IAX2->SIP |
ASTERISK-15450: Chanspy application does not exit when user hangs up |
ASTERISK-15451: No audio is passed from MOH when using originate to a remote peer |
ASTERISK-15452: [regression] Originate not launching secondary channel when primary is a Local channel |
ASTERISK-15453: [regression] Transfer is broken |
ASTERISK-15454: This is a test issue. There's nothing to see here. Please move along. |
ASTERISK-15455: This is a 2nd test issue. There's nothing to see here. Please move along. |
ASTERISK-15456: [patch] chan_misdn does not set INVALID_EXTEN |
ASTERISK-15457: asterisk crashes while fax sending |
ASTERISK-15458: macro-hangup executes after 15 minutes and 30 seconds on outbound calls |
ASTERISK-15459: MeetMe option 'x' is broken |
ASTERISK-15460: [patch] Dial option 'L' does not work correctly when a local channel is involved |
ASTERISK-15461: [patch] Update CDR variables that are available, before pbx starts |
ASTERISK-15462: Crash In chan_local in local_queue_frame (ast_mutex_trylock) |
ASTERISK-15463: [patch] Extend the max number of callgroups/pickupgroups |
ASTERISK-15464: [patch] main/feature: additional parking lots not reading needed variables |
ASTERISK-15465: [patch] Support for GROUP_MATCH_COUNT regex matching on category |
ASTERISK-15466: [patch] Setting "timerb" on chan_sip.conf doesn't work at all, in [general] or peer |
ASTERISK-15467: [patch] directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address |
ASTERISK-15468: [patch] Its not possible to pass more than one agrument in custom features. |
ASTERISK-15469: [patch] Session failure with specific SDP-Content (one media specific c= line, no session specific c= line) |
ASTERISK-15470: [regression] CDR attended transfer missing |
ASTERISK-15471: Asterisk don't update LDAP user's status |
ASTERISK-15472: If the UAC execute a SIP registration or deregistration, the LDAP settings don't change. |
ASTERISK-15473: [Asterisk 1.6.0-rc 6 update LDAP entries with "null" values] |
ASTERISK-15474: Possible problem with IAXVAR |
ASTERISK-15475: [patch] Double fields in SQL query |
ASTERISK-15476: Segfault under 1.4.23.2 |
ASTERISK-15477: Transfer hear silence when transfered is busy |
ASTERISK-15478: [patch] func_math MATH off by one's |
ASTERISK-15479: [patch] [regression] [patch] chan_sip does not check other mailboxes on AST_EVENT_MWI |
ASTERISK-15480: [patch] DISA doesn't honor caller ID on the channel |
ASTERISK-15481: Getting kernel: asterisk[4278]: segfault at 40 ip 006e6626 sp b70d7e38 error 6 in libc-2.9.so[66d000+16e000] |
ASTERISK-15482: RTP Timeout is flawed |
ASTERISK-15483: [patch] Random DTMF duplicate emulation on bridged OOH323 channel on outgoing calls |
ASTERISK-15484: DTMF not detected at all from Sipgate despite TCPDump showing keypresses |
ASTERISK-15485: [patch] Check on ac_cv_pthread_once_needsbraces fails |
ASTERISK-15486: ms sql connections problems |
ASTERISK-15487: Asterisk 1.6.0-rc 6 update LDAP entries with "null" values |
ASTERISK-15488: callee chanel overwrites the caller cdr |
ASTERISK-15489: Asterisk manage forked calls as reinvite |
ASTERISK-15490: [patch] TLS socket file descriptor fails to open (with no error message in log) |
ASTERISK-15491: [patch] channels stuck in ringing state forever |
ASTERISK-15492: "sip show peer/user <tab>" doesn't complete correctly |
ASTERISK-15493: 1.6.2.1: sip BYE issued 90 seconds into call |
ASTERISK-15494: [patch] deadlock in app_queue with use_weight during reload |
ASTERISK-15495: [patch] segfault on chanspy due to race in main/channel.c |
ASTERISK-15496: [patch] res_phoneprov.so causes Asterisk to crash on ${MAC}-phone.cfg file |
ASTERISK-15497: [regression] .call file not connecting to context: when channel: answers |
ASTERISK-15498: Hundreds (thousands?) of WARNING messages when data sent via res_phoneprov |
ASTERISK-15499: [patch] warning about "Invalid peer port configuration" for realtime |
ASTERISK-15500: [patch] AEL2 parser messages missing a final \n |
ASTERISK-15501: [patch] Configured CFLAGS/LDFLAGS are used by main, but not by modules built out of tree |
ASTERISK-15502: [patch] app_dial does not respect GOSUB_RESULT |
ASTERISK-15503: [patch] app_dial gosub does not pass back GOSUB_RETVAL |
ASTERISK-15504: [patch] Console documentation not loaded from XML |
ASTERISK-15505: [patch] Crash in res_agi when trying to send application usage |
ASTERISK-15506: chan_mobile crashes asterisk segmentatio fault on end of outgoing call |
ASTERISK-15507: [patch] new feature T.38 switch on/off from dialplan |
ASTERISK-15508: [patch] Missing fallback to audio fax feature when T.38 re-INVITE failed for 1.4 |
ASTERISK-15509: [patch] There is an Active call, even though device is Unregistered from asterisk! |
ASTERISK-15510: [regression] Voicemail admin only records .wav and .gsm, not .WAV greetings/unavailable/temporary messages |
ASTERISK-15511: [patch] New AgentTransfer manager event with extended transfer information |
ASTERISK-15512: [patch] Solaris sed fails on generating ael_lex.c |
ASTERISK-15513: IAX always attempts authentication against first (alphabetically) user |
ASTERISK-15514: gotoiftime does not work as expected for date range |
ASTERISK-15515: Segmentation Fault |
ASTERISK-15516: [regression] Attended transfer broken in 1.6.1.13 |
ASTERISK-15517: T.38 fix missing |
ASTERISK-15518: Lags when using imap |
ASTERISK-15519: [patch] [regression] 1.6.2.7 hangs during initial module load on Darwin |
ASTERISK-15520: [patch] [regression] T.38 negotiation Broken |
ASTERISK-15521: SIP URIs are not always parsed correctly |
ASTERISK-15522: [patch] Caller name lost during call redirect |
ASTERISK-15523: [patch] Introduce function for parsing ABNF name-andor-addr = name-addr / addr-spec |
ASTERISK-15524: [patch] Added a config parameter to report span and/or channels alarms using AMI |
ASTERISK-15525: [patch] [regression] DTMF Relaying appers broken |
ASTERISK-15526: Billing Differences between Carrier and Asterisk |
ASTERISK-15527: [patch] CHANNEL function cannot set OSP token for outbound IAX calls. |
ASTERISK-15528: [patch] Build issues with FreeBSD 6/8 |
ASTERISK-15529: [patch] PRI locks randomly, hangup cause 102, "recovery on timer expiry". |
ASTERISK-15530: 'interval' option doesn't work |
ASTERISK-15531: Orinate calls using AMI on version 1.4.29 broken. |
ASTERISK-15532: Unable to link Voicemail to voicemail accounts created using MySQL |
ASTERISK-15533: realtime oracle engine |
ASTERISK-15534: Every time I type "odbc show" it crashes in the next few seconds |
ASTERISK-15535: after a few minutes it takes down the server |
ASTERISK-15536: [patch] app_queue: Give members a penalty time for not answering |
ASTERISK-15537: type=user and type=friend are no longer the same for chan_sip |
ASTERISK-15538: coredump on T.38 Session with 1.6.2.1 |
ASTERISK-15539: [patch] Add support for configurable peer username in digest authentication |
ASTERISK-15540: I can't store CDRs in mysql DB |
ASTERISK-15541: Timezone (tz) parameter won't apply for users.conf |
ASTERISK-15542: [regression] Local channels broken for Originate and .callfiles : Call failed to go through, reason (3) Remote end Ringing |
ASTERISK-15543: configure --with-netsnmp fails if no openssl-devel |
ASTERISK-15544: [patch] core dump when user parkandannouce |
ASTERISK-15545: [patch] Add AMI support for device states |
ASTERISK-15546: Setvar DEVICE_STATE manager action occasionally ignored. |
ASTERISK-15547: Core Dump on Exit - v1.6.2/FreeBSD |
ASTERISK-15548: [patch] realtime oracle engine |
ASTERISK-15549: Callcentric Dropping Registration in Version 1.6.2 [see issue 0012312 ] |
ASTERISK-15550: large memory leak - ast_rtp_destroy not being call in all circumstances/other associated memory not being freed from chan_sip |
ASTERISK-15551: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.2.37.tar.gz does not contain issue# 0015765 |
ASTERISK-15552: [patch] New mute feature for MixMonitor |
ASTERISK-15553: Application Directory does not respect the [vm-context] context and selects users from the default context |
ASTERISK-15554: [patch] Missing description of the PARKINGLOT variable in XML documentation |
ASTERISK-15555: [patch] 'moh reload' doesn't reload moh directory content |
ASTERISK-15556: BASE64_DECODE broken again |
ASTERISK-15557: Caller ID info is destroyed after FXS channel is ringed |
ASTERISK-15558: Inbound calls are dropped after 15 mins and several Status 400 and 422 messages in SIP trunk against Huawei SoftX3000 |
ASTERISK-15559: [patch] DSP progress detection unable to detect SIT |
ASTERISK-15560: Audio loop reports T38 switchover but t38state != T38_STATE_NEGOTIATED |
ASTERISK-15561: [patch] Initial pause not implemented, but documented as available. |
ASTERISK-15562: IAX2 Reject Not Shown in Debug |
ASTERISK-15563: [patch] Clean transmit_* for start/stop media transmission |
ASTERISK-15564: [patch] Multiple segfaults in leave_voicemail at app_voicemail.c:4451 Asterisk 1.4.29 |
ASTERISK-15565: E1 PRI channel 'glare', where asterisk hangups up the inbound call from network. |
ASTERISK-15566: app_fax doesn't receive fax with T.38 |
ASTERISK-15567: [patch] Parking a call, then retrieving it with ParkedCall() kills the ability to transfer the retrieved call. |
ASTERISK-15568: [patch] zero/empty argument to gosub yields callers $ARG1 |
ASTERISK-15569: Attended transfers get incorrect voicemail. |
ASTERISK-15570: [patch] Adding manager event JabberStatus |
ASTERISK-15571: Sip Channels Colapse |
ASTERISK-15572: [patch] SIP call documentation - feel free to edit |
ASTERISK-15573: [patch] T.38 negotiation fails with Patton SN2400 |
ASTERISK-15574: [patch] Deadlock between handle_request_do and do_devstate_changes |
ASTERISK-15575: SIP Message parameters and URI parameters not parsed correctly |
ASTERISK-15576: [patch] Send manager event on AMI command Bridge |
ASTERISK-15577: [patch] Unrecognized prilocaldialplan NPI modifier |
ASTERISK-15578: T.38 with devices behind NAT does not work |
ASTERISK-15579: [patch] After AMI Bridge action the callerid's on the phones are not updated. |
ASTERISK-15580: Background application does not return until file is finished being played (Asterisk 1.6.0.10) |
ASTERISK-15581: [patch] [regression] 1.6.1.13 and 1.6.1.14 UDP ports not freed |
ASTERISK-15582: Queue with autofill=no and strategy=ringall sometimes rings non-oldest caller through to agents |
ASTERISK-15583: app_queue does not change member state |
ASTERISK-15584: [patch] Calendar SIGSEGV with iCal |
ASTERISK-15585: Calls retrieved from parking lot not recorded |
ASTERISK-15586: [patch] Pause After call |
ASTERISK-15587: make install fails when GNU install missing |
ASTERISK-15588: [patch] res_pktccops.so doesn't export a symbol, chan_mgcp will not load or will malfunction depending on gcc version |
ASTERISK-15589: [patch] 99.9 cpu when asterisk started with init.d script |
ASTERISK-15590: Park ed call slot annoucement is not heared |
ASTERISK-15591: [patch] Remove coloring escape sequences from log files. |
ASTERISK-15592: Asterisk drop calls sending a CSeq: 103 BYE |
ASTERISK-15593: [regression] Realtime agents Device state always Not in use |
ASTERISK-15594: [patch] Overlap receiving timeout, plus dialplan latency, causes network to retry SETUP |
ASTERISK-15595: [patch] 606 Not Acceptable is also a valid response to reject a T.38 re-INVITE |
ASTERISK-15596: T.38 session fails with '488 Not acceptable here' if within 5 seconds there is no "SIP 100 Trying" REINVITE reply from remote |
ASTERISK-15597: 1.4 does not send any SIP messages after the "100 Trying" to the T.38 INVITE requesting side |
ASTERISK-15598: Crash when destroying sip channels |
ASTERISK-15599: Originating a call from within the dialplan using Originate() does not result in a CDR |
ASTERISK-15600: [patch] Update DUNDi to XML docs |
ASTERISK-15601: astobj2.c:279 |
ASTERISK-15602: problems with high call frequency and more than 300 calls at the same time |
ASTERISK-15603: [patch] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory |
ASTERISK-15604: No Branch 1.6.1 Patch for Issue #0016766 |
ASTERISK-15605: [patch] ParkAndAnnounce Core dumps asterisk |
ASTERISK-15606: [patch] segfault in pri_schedule_del at prisched.c:124 |
ASTERISK-15607: [patch] make sounds doesn't download but make install does |
ASTERISK-15608: Create example documentation for usage of FILTER() to help secure dialplans |
ASTERISK-15609: [patch] Design functionality test for dialplan pattern matching |
ASTERISK-15610: Dialplan language does not deal with & character safely |
ASTERISK-15611: Heavy locking in manager.c results in eventual crash and loss of CLI commands and intense CPU load |
ASTERISK-15612: dfgfdg |
ASTERISK-15613: Failure of canary |
ASTERISK-15614: [patch] chan_sip does not decrease module refcount on deferred BYE |
ASTERISK-15615: [patch] Attended transfer broken in 1.6.2.2 |
ASTERISK-15616: [patch] Reload command does not update the SLA configuration properly |
ASTERISK-15617: [patch] Manager event AgentComplete should alway be sent |
ASTERISK-15618: Parameter m in Dial command |
ASTERISK-15619: SDP in a session refresh re-INVITE does not contain T.38 when it should |
ASTERISK-15620: IAX2 queue member Unknow instead of Not in Use |
ASTERISK-15621: (Version 1.4.29) Queue with autofill=no and strategy=ringall sometimes rings non-oldest caller through to agents |
ASTERISK-15622: [patch] 603 Declined when call torn down |
ASTERISK-15623: Discrepancy among core-sounds-en.txt and sounds in 1.4.17 set |
ASTERISK-15624: [patch] Endianess problems in skinny messages |
ASTERISK-15625: Asterisk does not honor the bindport. |
ASTERISK-15626: [patch] DEBUG_THREADS - Compile errors on OS/X Snow Leopard |
ASTERISK-15627: Audio pauses at the same time as rtcp report is handled |
ASTERISK-15628: [patch] Build fails on smsq.c with syntax error |
ASTERISK-15629: Random crashes |
ASTERISK-15630: [patch] [regression] autofill=no always IGNORED. |
ASTERISK-15631: Asterisk 1.6.2.3 RC2 CRASHING RANDOMLY |
ASTERISK-15632: seg fault in _ast_calloc at utils.h:462 |
ASTERISK-15633: [patch] core show sysinfo shows invalid (negative value) for ram on systems whith a lot of ram |
ASTERISK-15634: freeplay and opsound tarballs are identical |
ASTERISK-15635: ASterisk 1.6.0.13 |
ASTERISK-15636: [patch] Deadlock in chan_local when obtaining locks on local_pvt->lock |
ASTERISK-15637: Codec translation path builder does not produce expected results with 16khz and 32khz audio |
ASTERISK-15638: [patch] [regression] system() dialplan function fails (-1 returned) when argument is single quoted |
ASTERISK-15639: [patch] Fix portability bit fields and make "mfcr2_immediate_accept" work again |
ASTERISK-15640: Incorrect linker flags used on OpenSolaris |
ASTERISK-15641: Unable to create channel for non registered SIP devices |
ASTERISK-15642: [patch] Deadlock between dahdi_exception and dahdi_indicate |
ASTERISK-15643: [patch] Voicemail attachments are sent even with attach=no |
ASTERISK-15644: Should there be transcoding after attended transfer? |
ASTERISK-15645: CRASH ON 1.6.2.3 RC2 |
ASTERISK-15646: [patch] Some issues with New SDP handling code and T.38 |
ASTERISK-15647: [patch] Memory leak in realtime meetme |
ASTERISK-15648: [regression] Crash in app_voicemail.c in function inprocess_cmp_fn becuase j->context is NULL |
ASTERISK-15649: Template examples in documentation imply well defined overriding semantics, but this is not true |
ASTERISK-15650: [regression] Blind transfers initiated from calling party aren't disconncted |
ASTERISK-15651: Incorrect checking of Refer-To and Referred-By SIP headers |
ASTERISK-15652: [patch] asterisk command history loads as unusable garbage |
ASTERISK-15653: Music on hold broken in 1.6.2.2 |
ASTERISK-15654: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
ASTERISK-15655: [patch] [regression] One-legged Transfer (INVITE / Replaces) not working anymore |
ASTERISK-15656: [patch] PickupChan is not working |
ASTERISK-15657: [patch] Voicemail minsecs is not able to be overridden per-mailbox |
ASTERISK-15658: [regression] No Ringing / No Playback when call is getting forwared by a phone |
ASTERISK-15659: [patch] Set conterence options on pin-less RealTime conferences |
ASTERISK-15660: Phone generated dial tone not switched off during call |
ASTERISK-15661: One way audio after placing call on hold and resuming |
ASTERISK-15662: [patch] Make verb and noun declination available in dial plan |
ASTERISK-15663: [regression] AGI function GET DATA does not play audio file |
ASTERISK-15664: codec_dahdi.c gives error message on startup after fresh install |
ASTERISK-15665: Calltoken's and IAX2 realtime configuration |
ASTERISK-15666: [patch] Cleanup transmit_* functions |
ASTERISK-15667: [patch][regression] Cannot specify http:// URL in music-on-hold class directory statement |
ASTERISK-15668: setting pollmailboxes=yes will crash asterisk on startup |
ASTERISK-15669: ael context extend does not look at macros |
ASTERISK-15670: [patch] Cleanup transmit_displaymessage |
ASTERISK-15671: [patch] Enable opening an audio stream post-startup |
ASTERISK-15672: [patch] unable to exit echo application with '#' |
ASTERISK-15673: Asterisk sends voicemails to a second totally wrong receiver |
ASTERISK-15674: Avaya IP Office 5.0 probably crashes Asterisk |
ASTERISK-15675: Asterisk 1.6.1.16 crashes randomly |
ASTERISK-15676: [patch] CLI command 'core show codecs' does not display slin16 codec 0x8000 |
ASTERISK-15677: Attended transfer is broken on 1.6.2.4 |
ASTERISK-15678: /etc/init.d/asterisk.debian consumes 100% CPU |
ASTERISK-15679: Sip Channels Colapse - Too much sip channels |
ASTERISK-15680: Asterisk 1.2.39 core dump |
ASTERISK-15681: [regression] Context option in queue.conf no longer works |
ASTERISK-15682: IAX2 crash's randomly |
ASTERISK-15683: queuelog does not show correct extension on transfers using non Local/ members |
ASTERISK-15684: [patch] Realtime queue does not re-read announce variable from mysql after first use |
ASTERISK-15685: Repark a call on Parkinglot |
ASTERISK-15686: Asterisk Crashes after it thinks it gets corrupt SIP message |
ASTERISK-15687: [patch] From-header parsed twice for each invite or subscription request |
ASTERISK-15688: dnsmgr failes to match peer when SIP srvlookup is on |
ASTERISK-15689: srvlookup don't work with register |
ASTERISK-15690: [patch] patch for LISTFILTER to properly handle delimiter |
ASTERISK-15691: Asterisk 1.4.29 musiconhold of remote party doesnt work |
ASTERISK-15692: Compile fails in dahdi_show_status with kernel 2.6.33 |
ASTERISK-15693: [patch] Incorrect pattern specificity in new dial pattern functions |
ASTERISK-15694: [patch] [regression] Duplicate TXREQ packets will cause chan_iax2 to reject an unrelated call in the future |
ASTERISK-15695: [patch] system() dialplan function does not work |
ASTERISK-15696: [patch] moh files install under datadir, at runtime: under varlibdir |
ASTERISK-15697: jerky audio after a while |
ASTERISK-15698: [patch] SIP autocreate peers registered when request to unregister |
ASTERISK-15699: [patch] Useful new wildcards to ease secure dialplans |
ASTERISK-15700: [patch] Alignment trap on ARM processor on calculating cost of codec |
ASTERISK-15701: crash in musiconhold |
ASTERISK-15702: Adaptive jitterbuffer causes 30 seconds of no audio. |
ASTERISK-15703: jblog=yes does not create jblog like expected |
ASTERISK-15704: regcontext not handled in a fashion similar to chan_sip |
ASTERISK-15705: Allow regcontext per peer |
ASTERISK-15706: [patch] VoiceMail(vmbox@context,s) -> Regularly segfaults asterisk |
ASTERISK-15707: Duplicates of uniqueID |
ASTERISK-15708: RTP traffic only seen in one direction when using FollowMe() |
ASTERISK-15709: t38pt_usertpsource=yes seems to work incorrectly with ReceiveFax |
ASTERISK-15710: [patch] app_queue: Log failed attempts to call members |
ASTERISK-15711: Add support for reporting/passing all CallerID data to app_queue - specifically RDNIS,DNID |
ASTERISK-15712: [patch] Segfault branches, and trunk, when DAHDI FXS port goes off hook |
ASTERISK-15713: [patch] The Dial c option returns answered elsewhere if the dial timeout occurs (only tested using SIP) |
ASTERISK-15714: [patch] Patch to fix 15609 broke followme |
ASTERISK-15715: [patch] [regression] app_followme playing wrong sound files |
ASTERISK-15716: mutual exclusion with optiom m and L in dial |
ASTERISK-15717: unexpected end of a call for 20-25 seconds |
ASTERISK-15718: [patch] [sounds] Two additional sound prompts for use with ConfBridge |
ASTERISK-15719: glibc 2.11.1 causes asterisk to not start due to return code from dlclose |
ASTERISK-15720: Buggy parse of request-line in function check_user_full() |
ASTERISK-15721: [patch] Qualify frequency has big pauses. Asterisk stops sending SIP OPTIONS to keep NAT alive |
ASTERISK-15722: For loop never exists when calling an extension that exists with BUSY, failure to leave 'for' loop in 'scan_thread' |
ASTERISK-15723: [patch] Cleanup transmit_callstate handling |
ASTERISK-15724: [patch] Problem inserting CDR records when certain characters are used |
ASTERISK-15725: SIP RTP audio delay |
ASTERISK-15726: [patch] chan_mgcp crash Adtran 624 asterisk 1.8.0 Beta 2 |
ASTERISK-15727: 100% CPU load at feature startup |
ASTERISK-15728: Need to port a feature from Trunk to 1.6.X |
ASTERISK-15729: Monitor() does not handle filenames with path correct |
ASTERISK-15730: [patch] fix getting callerid name in imap_retrieve_file() (broken callerid number announcement/reply/...) |
ASTERISK-15731: Call that clears in same app_dial poll as answer is reported as NOANSWER but NORMAL_CLEARING |
ASTERISK-15732: app_fax exits with returncode -1, transmission error, although fax is sent correctly, AGI aborts execution |
ASTERISK-15733: Callerid Channel dahdi port FXS hook up |
ASTERISK-15734: sip reinvite broken |
ASTERISK-15735: All extensions are patterns (despite it does not begin with _ (underscore) if extenpatternmatchnew=yes |
ASTERISK-15736: It crashes on func_odbc |
ASTERISK-15737: [patch] [regression] SQL Syntax Error - Missing Single Quote |
ASTERISK-15738: [patch] [regression] local channel with '/bn' modifier does not optimize itself on built-in transfer |
ASTERISK-15739: In dialplan (extensions.conf), subdirectories are not respected |
ASTERISK-15740: [patch] Rogue Newchannel events for failed Originate calls |
ASTERISK-15741: crash in main/cli.c: find_cli |
ASTERISK-15742: SIP-Provider without "SIP/2.0 180 Ringing" makes trouble with call file |
ASTERISK-15743: Read factory 0xb6d0acb8 was pretty quick last time, waiting for them |
ASTERISK-15744: UserEvent Documentation is incorrect |
ASTERISK-15745: [patch] Automatic add UniqueID to user event |
ASTERISK-15746: [patch] Update to new local channel documentation |
ASTERISK-15747: When transfering call, sound disappear. |
ASTERISK-15748: [patch] DBGet response does not end with a 'Complete' event |
ASTERISK-15749: SpeechBackground with multiple files does not interrupt speech when DTMF is received |
ASTERISK-15750: Nested Dial()s that use U() or M() results in: '&(audiohook)->lock' freed more times than we've locked! |
ASTERISK-15751: [patch] Callerid Channel dahdi port FXS are empty after the first hangup. |
ASTERISK-15752: Catch privateNumberDigits from h323 pdu |
ASTERISK-15753: Using func_odbc.conf |
ASTERISK-15754: When playing from ExternalIVR, the playback is very fast (about double the speed of standard Playback) |
ASTERISK-15755: [patch][feature] Add device state capabilities to ConfBridge (similar to MeetMe) |
ASTERISK-15756: multiple values for "dtmfmode=" per trunk |
ASTERISK-15757: Deadlocks with ~2k MGCP users |
ASTERISK-15758: [patch] OOH323 connection to Avaya IPOffice 403 drops i n 394 seconds |
ASTERISK-15759: After upgrade from 1.4.21 to 1.4.29 on internal SIP calls don't hear the ringback tone |
ASTERISK-15760: cli help for alias doesnt do anything sensible |
ASTERISK-15761: contributed init.d file has wrong pid location |
ASTERISK-15762: [patch] Cleanup transmit_ for handle_register and keepalives |
ASTERISK-15763: [patch] MusicOnHold ignores realtime when no musiconhold.conf classes configured. |
ASTERISK-15764: alarm state not properly maintained on analog channels |
ASTERISK-15765: impossible to pass more than one argument to a custom applicationmap in features.conf |
ASTERISK-15766: Asterisk segmentation fault |
ASTERISK-15767: ${VOICEMAIL_EXTEN} doesn't get set at asterisk startup |
ASTERISK-15768: [patch] small error in T.140 RTP port verbose |
ASTERISK-15769: When attempting to dial specific number pattersn a CHANUNAVAIL message is received |
ASTERISK-15770: When Answer is used for chan_local in Originate, the originate go crazy |
ASTERISK-15771: asterisk 1.6.2.6-rc2 has core show locks where DEBUG_CHANNEL_LOCKS is not set |
ASTERISK-15772: [patch] Can't build asterisk using just --with-netsnmp... seems to always want to use CONFIG_NETSNMP |
ASTERISK-15773: incoming INVITE received no progress, just 200 OK, causing Sipra pstn to go off-hook |
ASTERISK-15774: [patch] MusicOnHold produces a crash |
ASTERISK-15775: [patch] Cleanup transmit_* functions |
ASTERISK-15776: [patch] Crash in app_voicemail.c in function retrieve_file (Read out in small chunks) |
ASTERISK-15777: Audio problems in Meetme when DAHDI channel is Monitored |
ASTERISK-15778: [patch] Chan_sip compile never completes on Mac OSX 10.6.2 |
ASTERISK-15779: Transfer fails |
ASTERISK-15780: [patch] t38pt_udptl=no on a peer isn't respected (can't disable t38 negotiation on a per-peer basis) |
ASTERISK-15781: Bug in Calculating T.38 far max IFP? |
ASTERISK-15782: [patch] [regression] Segfault when hanging up phone after launching app_confbridge on Solaris 10 x86 |
ASTERISK-15783: [patch] sqlite module does bad parsing of values in config file |
ASTERISK-15784: Crash when using an SQL Server ODBC connection (FreeTDS) |
ASTERISK-15785: [regression] Set(CDR(mycol)=xxx) does not get populated in the backend when the CDR is posted |
ASTERISK-15786: Data Buffer Size Exceeded! |
ASTERISK-15787: [patch] Asterisk sends session-timer with "require" after 15 minutes |
ASTERISK-15788: [patch] func_odbc query is limited to 15 characters |
ASTERISK-15789: [patch] trivial patch to notifiy CLI when module loaded / unloaded. |
ASTERISK-15790: [patch] RTP Timestamp changes after transfer, but SSRC not and the markerbit ist not set. |
ASTERISK-15791: [patch] Autopause only pauses member on queue where timeout took place. |
ASTERISK-15792: Dialplan continues execution after transfer |
ASTERISK-15793: When a context is defined in [general] section of sip.conf, other contexts are ignored |
ASTERISK-15794: Context statement removed from [general] section of sip.conf, but remained active even after a reload |
ASTERISK-15795: [patch] endless wait for RTP in certain scenarios |
ASTERISK-15796: ./configure -help describes wrong default behavior |
ASTERISK-15797: Can't receive Fax with Patton 4554 ISDN Gateway and SPA 2102 VoIP Adapter |
ASTERISK-15798: Saturated Handles, Socket Error |
ASTERISK-15799: [patch] [regression] videosupport=always acts like videosupport=no |
ASTERISK-15800: SQLite3-3.6.23 incompatibility in asterisk-1.6.2.6 |
ASTERISK-15801: [patch] Background application not use 'context' parameter |
ASTERISK-15802: SIP response 415 "Unsupported Media Type" when using G729 |
ASTERISK-15803: On omitting the T flag from Dial() the caller can still make a blind transfer |
ASTERISK-15804: [patch] Exchange Web Service calendaring support |
ASTERISK-15805: [regression] Manager Events Exit Early |
ASTERISK-15806: Park is passed incorrect parameters if a Dial application is executed in a subroutine anywhere on the system. |
ASTERISK-15807: [patch] This is C. Indent levels do not matter in C. |
ASTERISK-15808: Asterisk crashes with app_fax / spandsp when compiled in 64 bit |
ASTERISK-15809: [patch] Makefile update to only build asterisk.conf |
ASTERISK-15810: [patch] make clean: /bin/sh: /usr/bin/sw_vers: not found |
ASTERISK-15811: 181 forwarded messages incorrectly interpreted |
ASTERISK-15812: Crash with 'sip reload' when system has some load |
ASTERISK-15813: [patch] Makefile: remove ASTBINDIR variable |
ASTERISK-15814: When using originate Local/.../n, dest extension does not run on Local channel pickup |
ASTERISK-15815: Jabber Module, crash and/or no authentication |
ASTERISK-15816: [patch] Problems with MeetMe and RT schedule dates |
ASTERISK-15817: Conflict in sample configurations with TRUNK, etc. |
ASTERISK-15818: [patch] Perl script to import CDR text file to ODBC database table |
ASTERISK-15819: [patch] internal_ao2_ref fails to check if null returned from INTERNAL_OBJ |
ASTERISK-15820: [patch] Insert fails when database initialized during connection outage to postgres server |
ASTERISK-15821: [patch] allow using system copy of libedit |
ASTERISK-15822: [patch] Explicit context set in SIP peer overridden by default domain context |
ASTERISK-15823: [patch] Documentation for configuring asterisk faxdetect for fax to email with spandsp |
ASTERISK-15824: Create CDR queue so records are not lost when connectivity is lost |
ASTERISK-15825: dtmf logging to console not working |
ASTERISK-15826: [regression][patch] SDP c and o lines contain the wrong IP address when using an externally mapped IP(extern{ip,host}) |
ASTERISK-15827: Configure script inconsistent in using CFLAGS when detecting header files |
ASTERISK-15828: [patch] Add new AGI command: PARK |
ASTERISK-15829: [regression] Some AMI Originate Calls on chan_local fail/timeout |
ASTERISK-15830: [regression] Incoming DAHDI to DAHDI bridged channels staying with ringing status |
ASTERISK-15831: [patch] CLI commands via asterisk -rx may not return all output |
ASTERISK-15832: channel.c:2753 __ast_read: Exception flag set on 'SIP/XXXXXXXX-000000e3', but no exception handler |
ASTERISK-15833: main/test.c reports erroneous cli message |
ASTERISK-15834: chanspy on channel in MeetMe conference results in a crash |
ASTERISK-15835: Asterisk crash - core dump at manager.c:2637 |
ASTERISK-15836: strange documentation of tlsbindaddr in sip.conf |
ASTERISK-15837: timeout problem in queues |
ASTERISK-15838: [patch] Clear received caller ID number and name on DADHI hangup |
ASTERISK-15839: [patch] peer section does not allow to configure a port |
ASTERISK-15840: [patch] Freenum-in-a-can configuration for configs/extensions.conf.sample |
ASTERISK-15841: What causes 'Bad Magic Number'? As immediately after segfault. |
ASTERISK-15842: problem with announce frequence in queues |
ASTERISK-15843: Queue hangup if periodic announcement file is missing |
ASTERISK-15844: [patch] Meaningless extension warnings logging |
ASTERISK-15845: CPU usage increases if WaitEvent not called |
ASTERISK-15846: redirect to fax extension only after ring |
ASTERISK-15847: [patch] Real-time Priory |
ASTERISK-15848: Adaptive Jitter Buffer issue |
ASTERISK-15849: [patch] chan_dahdi/fxs really needringing |
ASTERISK-15850: AGI->wait_for_digit or AGI->exec('Read' do not report digits back on an outgoing call |
ASTERISK-15851: asterisk is not closing unused RTP ports |
ASTERISK-15852: When using another SIP Trunk, Asterisk generates the initial ring RING as a response to "SIP SESSION PROGRESS" |
ASTERISK-15853: Asterisk 1.6.1.18 and 1.6.2.6 RTP BUG |
ASTERISK-15854: [patch] Added response timeout option to SendFax() appliation |
ASTERISK-15855: No ENTERQUEUE event in queue_log if leavewhenempty=yes in queues, therefore no ACD report is available to track overflow calls. |
ASTERISK-15856: Unable to Install Asterisk in asterisk-1.6.0.21 |
ASTERISK-15857: [patch] [regression] fix for #16802 forces change of astrundir ownership, breaking socket perms |
ASTERISK-15858: [patch] Fix query with double backslash in string literals and stop log warnings |
ASTERISK-15859: [patch] MixMonitor records shorter files than the call duration. |
ASTERISK-15860: qsigchannelmapping parameter |
ASTERISK-15861: [patch] Asterisk crashes while core restart (#0 0x000000000050683c in term_beep (el=0x16cdd9b0) at term.c:865) |
ASTERISK-15862: [patch] Memory Leak in app_queue |
ASTERISK-15863: [patch] Improve realtime queue logging |
ASTERISK-15864: [patch] Asterisk crash in func_odbc |
ASTERISK-15865: [patch] [regression] Overlap dialing to PSTN failing after #16789 |
ASTERISK-15866: [patch] 'core show settings' should show all settable directories |
ASTERISK-15867: [patch] Segfault in manager event after fax receipt |
ASTERISK-15868: app_followme + cdr_adaptive_odbc crashes when followme progresses from number set to number set |
ASTERISK-15869: Background behaves strangely[t when priority 1 is not available in current extension. |
ASTERISK-15870: MWI SIP NOTIFY may contain wrong "Via: ..." header, making the phone discard the whole message |
ASTERISK-15871: T.38 faxmaxdatagram overflows with UDPFEC, works with "t38pt_udptl=yes,redundancy" (udptl.c) |
ASTERISK-15872: [patch] Segmentation fault when using two codec modules that register the same src and dst format |
ASTERISK-15873: [patch] AGI SPEECH SET bugs |
ASTERISK-15874: unportable shell in safe_asterisk |
ASTERISK-15875: On receiving Fax with ReceiveFax Asterisk core dumps |
ASTERISK-15876: C keeps ringing when hanging A and B after blind transfer using atxfer |
ASTERISK-15877: [patch] Pickup with Aastra phones: Unable to find subscription |
ASTERISK-15878: Crash just after call is answered |
ASTERISK-15879: [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak |
ASTERISK-15880: AMI IAXpeers not "complete," no actionID |
ASTERISK-15881: [patch] Asterisk Crashes With Core Dump When FUNC_ODBC Processes SQL Statements With Errors |
ASTERISK-15882: [patch] [regression] No message is actually prepended when sending a voicemail message to another user |
ASTERISK-15883: 1.6.1.18 -> 1.6.2.6 T38 Fax: call drops |
ASTERISK-15884: No progress playing a sound after hangup |
ASTERISK-15885: Add a "pickupmacro=" setting in features.conf for execution on undirected pickup (typically *8) |
ASTERISK-15886: [patch] [regression] openh323 log is not printed on asterisk console |
ASTERISK-15887: [regression] realtime queues: persistent members not loaded from astdb on restart |
ASTERISK-15888: MeetMe 'useropts' and 'adminopts' not taken into account when RealTime is used |
ASTERISK-15889: Far Max IFP/Local max datagram Calculation and the reality |
ASTERISK-15890: ChannelRedirect() fails to properly redirect some of the time |
ASTERISK-15891: Segfault error |
ASTERISK-15892: When providing a delemeter character, placing it betwenn apostrophy signs does not work. CUT retruns the shole string in this ca |
ASTERISK-15893: Socket leaks when SIP call is rejected |
ASTERISK-15894: ChannelRedirect() fails to redirect |
ASTERISK-15895: Random crashes |
ASTERISK-15896: (Regression) Pickup from Grandstream BLF button ignores the context specified in Pickup command |
ASTERISK-15897: Music On Hold Not reloading the files using moh reload |
ASTERISK-15898: Using pattern match in a hint causes deadlock under described conditions |
ASTERISK-15899: [patch] Missing file queue-minute causes hangup of queue calls with wait time announcement = 1 minute |
ASTERISK-15900: [patch] Potential malfunction due to unitialized local variable |
ASTERISK-15901: [patch] minmemfree does not work |
ASTERISK-15902: [patch] Make transfer calls more pattern friendly |
ASTERISK-15903: [patch] Softkey redial with no previous number segfaults |
ASTERISK-15904: [patch] Wrong encoding of SIP URI |
ASTERISK-15905: [patch] endless cycle in ast_waitfor_nandfds() for big timeouts |
ASTERISK-15906: Asterisk crashes after "Stopped music on hold on DAHDI" (or tranfer to SIP channel may be) |
ASTERISK-15907: voicemail blasting to users with voicemail as an email does not seem to be working |
ASTERISK-15908: SENDFAX is not working for me |
ASTERISK-15909: Dial 'm' option produces a lot of warnings on DAHDI channel |
ASTERISK-15910: When single user in Meetme application there are scrolling errors from ast_dsp_silence |
ASTERISK-15911: Asterisk 1.4.30 crashes on transers with the patch around CONNECTEDLINE (https://issues.asterisk.org/view.php?id=8824#118065) |
ASTERISK-15912: [patch] Message count incorrect |
ASTERISK-15913: [patch] SegFault when connecting incoming call to parked call. |
ASTERISK-15914: Conference sound files re-recorded |
ASTERISK-15915: [patch] CallerID not properly set when using Originate and AGI |
ASTERISK-15916: app_voicemail crashes intermittently when voicemail box is over maxmsg |
ASTERISK-15917: Request to change Name/username field truncation from 11 to 12 chars |
ASTERISK-15918: [patch] Enable PRI SERVICE message support in chan_dahdi |
ASTERISK-15919: hidecalleridname parameter in chan_dahdi.conf |
ASTERISK-15920: [patch] Endpoints are not loaded when using Realtime |
ASTERISK-15921: AEL warning when using Return() application is misleading |
ASTERISK-15922: Extra lines in msg<number>.txt is added when forwarding a prepended mail to another mailbox |
ASTERISK-15923: Segfault with too many IMAP voicemails |
ASTERISK-15924: Everyone is busy/congested at this time |
ASTERISK-15925: Busy(xx) exits immediately on IAX channel |
ASTERISK-15926: ENUMQUERY fails for seemingly valid e164.org record |
ASTERISK-15927: Playback deadlock? |
ASTERISK-15928: [patch] CLI command logger set level auto complete |
ASTERISK-15929: chan_sip sends to peer mwi notify for wrong mailbox |
ASTERISK-15930: [patch] Add ${TOTALCALLS} dialplan variable |
ASTERISK-15931: SIP DTMF problem using RFC2833 between 1.2 <-> 1.4 <-> Unknown-brand/model SBC |
ASTERISK-15932: [patch] Command/Response queue stuck |
ASTERISK-15933: proxy_allocate() fails to init proxy->ip.sin_family |
ASTERISK-15934: MeetMe Options S(10)L(10000) |
ASTERISK-15935: [patch] HowTo: Collecting Debug Information |
ASTERISK-15936: [patch] CLI prompt interfers with CLI output |
ASTERISK-15937: [patch] OSARCH in GNU/kFreeBSD |
ASTERISK-15938: missing libs in link command of chan_h323.so: module fails to load |
ASTERISK-15939: [patch] Bluetooth linkkey is never stored and as a result phone PIN is requested every time a new connection is started |
ASTERISK-15940: [patch] ALL VERSIONS ! between two asterisks and peers authenticate as coincidental name invite |
ASTERISK-15941: [branch] update live_ast for testsuite |
ASTERISK-15942: Caller ID from internal DAHDI extensions not detected |
ASTERISK-15943: RECONNECT fails to work for Conference Call |
ASTERISK-15944: [patch] downgrade ast_debug from 1000 to 10. |
ASTERISK-15945: [patch] no sound on Playback(<file>,noanswer) |
ASTERISK-15946: Specifying return_context for ParkAndAnnounce affects 'h' exten behavior |
ASTERISK-15947: [patch] cdr_mongodb |
ASTERISK-15948: [patch] sip-friends.sql Missing 'useragent' |
ASTERISK-15949: [branch] Appdoc for manager events |
ASTERISK-15950: Connection Problem with UnixODBC 2.1.14 |
ASTERISK-15951: [patch] Updated Mantis Work Flow Documentation |
ASTERISK-15952: sip show channelstats isses. |
ASTERISK-15953: asterisk 1.4.23 regularly core dumps |
ASTERISK-15954: SDP does not get parsed when in SIP multipart body below line 64 |
ASTERISK-15955: [patch] SetCallerpres not honored on SIP Redirect |
ASTERISK-15956: [patch] Add a parameter to SendDTMF dialplan application |
ASTERISK-15957: [patch] Updates to Application Documentation |
ASTERISK-15958: [patch] [regression] Using Local channels with queues causes deadlocks |
ASTERISK-15959: [patch] OOH323 Outgoing Call Fails if Originated from a DAHDI Extension |
ASTERISK-15960: crash when calling ao2_unlock inside pthread_timer_disable_continuous - NOT FIXED PLEASE RE-OPEN |
ASTERISK-15961: lack of locking in dahdi_request() |
ASTERISK-15962: [patch] Update the doc/backtrace.txt documentation |
ASTERISK-15963: Conference will not record with DAHDI loaded, but no hardware |
ASTERISK-15964: [patch] Queue announcement playing times are delayed by retry+timeout seconds in slot 0 (1.4.26.2) |
ASTERISK-15965: [patch] Phone keeps ringing when hangup between 'NOTIFY' and 'Status: 180 Ringing' |
ASTERISK-15966: ast->tech_pvt->rtp contains garbage yielding SEGFAULT |
ASTERISK-15967: [patch] Suggestion to add Seconds on both cases of Action Status of Manager |
ASTERISK-15968: call distribiution problem in fewestcalls and lastrecent strategy |
ASTERISK-15969: Unable to exit Directory() application with large number of users (same last name) |
ASTERISK-15970: Changing storm-prevention behaviour in logger.conf |
ASTERISK-15971: Core dumped |
ASTERISK-15972: RecordFile API does not return after timeout |
ASTERISK-15973: Asterisk core dumps using MOH |
ASTERISK-15974: Over Threshold |
ASTERISK-15975: [patch] Dial()'s do_forward() breaks Local/ channel frame forwarding |
ASTERISK-15976: module app_voicemail not load |
ASTERISK-15977: Using SORT strings with comma in them are truncated up to the first comma. |
ASTERISK-15978: MOH+Madplay when going from one to another the second stream is not started |
ASTERISK-15979: Crash on Hangup during post_cdr |
ASTERISK-15980: Crash on Park execute |
ASTERISK-15981: Asterisk consumes 100% CPU, high interupt load, calls stay at ringing state |
ASTERISK-15982: Segmentation fault: blindxfer (#) using SIP |
ASTERISK-15983: Segmentation fault and restart asterisk |
ASTERISK-15984: Unabled to xfer call picked up from ParkedCall |
ASTERISK-15985: Exceptionally long voice queuing when using chan\Local to playback Extensions |
ASTERISK-15986: [patch] Segfault on ael parsing |
ASTERISK-15987: [patch] Deadlock between ast_hangup and pri_dchannel |
ASTERISK-15988: 100% CPU load on debian when loading from initscript |
ASTERISK-15989: All channels get Congestion following PTP MDL can't handle error of type F message |
ASTERISK-15990: [patch] Missing Menuselect Option "Compiler Flags - Development" in dev mode |
ASTERISK-15991: [patch] Add ability to generate an ASCII document from the TeX files |
ASTERISK-15992: [patch] Originate Action output is inconsistent with other manager actions |
ASTERISK-15993: [patch] Colourized logging option request as an option |
ASTERISK-15994: [patch] System() taking excessive time to return with nonroot |
ASTERISK-15995: Asterisk crashes sometimes (device state?) |
ASTERISK-15996: [patch] using opal instead of pwlib/openh323 |
ASTERISK-15997: [patch] Segmentation fault with unanswered inbound call via chan_ooh323 |
ASTERISK-15998: Asterisk 1.4.29 crashes in astobj2.c |
ASTERISK-15999: contexts and voicemail |